Digital audio signal watermarking in real-time is difficult in an environment that has limited processing power. This is for example the case on an embedded platform in which due to cost, heat and loudness reasons usually low power processing units are used, or in a server in which a powerful processor has to watermark in real-time several data streams in parallel.
Usually audio watermarking systems are operating in a block based manner where the watermark (WM) embedder gets a block of N input signal samples, WM processes this block and returns a block of N modified output signal samples. Real-time means that the time period available for WM processing of a signal data block is shorter than the time period used to get the next signal data block. If the WM processing time is longer, the real-time constraint is violated and a buffer overflow at the input of the embedder will occur, which leads to dropping of samples and audible artifacts and degradation of the audio quality.
In addition, the processing time required for watermark embedding is often audio signal content-dependent.